Download Pulsaret_2 user manual

Pulsaret
granular synthesis
u s e r m a n u a l v. 2 . 0 . 0
apeSoft
www.densitygs.com
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Starting
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system requirement
installation
copyright
demo limitation
purchase a license and authorizes
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update to v.2
background
GUI layout
DSP settings
Mixer
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faders and solo/mute
I/O mapping
monitoring
on/off stream trigger
• native effects
• VST slots
• Recorder
Filter Equalizer
general
compilers
cutoff/gain
Fibonacci COMPILER
• Geometric COMPILER
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• Harmonic COMPILER
• Scalar COMPILER
• transition subscribe
Granular streams
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parameters
settings
grain envelope
wave select
Windowing
• general
• shape select
• draw
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• additive
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audio slot
WavePad
WavePad display
WavePad Snap
WavePad management
Snapshots
• store/recall
• micro pad
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transitions
snapshots manage
rename window
subscribe/un-subscribe transition clients
transition curve
Snapshots sequencer improviser unit
• beats cycle
• time intervals
• step sequencer
• output
• global transport
Hyper Vectorial Pad
• general explanation
• pad mouse behavior
• X/Y time
• miscellaneous
Matrix
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general explanation
parameter linkage
grailet/pulsaret mode
parameters rescale
• LFO modulation
• status bar
MIDI/OSC I/O
• header
• sync
• widget mapping
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• stream select
• shared
• pick-up
Overview
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save/load project
project tree
status bar
clue window
float window
• history
• acknowledgments
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Starting
• system requirement
Macintosh
Pulsaret requires a Mac PPC or Intel machine running OS X 10.4 or later, and 1 GB RAM.
Windows
Pulsaret requires a Windows XP/Vista/7 machine and 1 GB RAM.
Pulsaret uses QuickTime in order to read correctly media files (including MP3), therefore QuickTime must be
installed on your system. On Windows we recommend a complete installation of QuickTime choosing all
optional components.
• installation
Macintosh
1. Download Pulsaret.dmg from www.densitygs.com
2. Double click to open the disk image
Windows
1. Download Pulsaret_setup.msi from www.densitygs.com
2. Double click to install
• copyright
This program is copyright shareware and it is not freeware.
You can download the unregistered version of the program and give it to your friends or to any other person
as long as for no charge. This program cannot be distributed in shareware compilations CDs without prior
written approval from the author.
No responsibility is taken for any damage or losses caused by this package.
All program trademarks belongs to its respective author.
• demo limitation
Demo version it is fully functional but the application will quit after ten minutes. One every minute Pulsaret
emit a white-noise. Buying a license you will disable this hateful behavior.
• purchase a license and authorizes
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Download the software from the web page: www.densitygs.com or somewhere else:
Purchase the license software from Kagi: http://store.kagi.com/?6FHML_LIVE&lang=en
in five/ten minutes you will receive an email with your activation code;
Launch the software, open the “about” then click Authorize button. Now you need enter activation code,
sent you from Kagi (Thanks for your purchase e-mail) and your e-mail employed for the kagi transaction.
Click on Authorize to register you license.
• update to v.2
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You can UPDATE to Pulsaret v.2 from Pulsaret v.1 (Mac/Win). N.B. you must already own a Pulsaret v.1
registration code! Buy Pulsaret v.2 from Kagì by inserting UPDATE COUPON DISCOUNT sent you from
apeSoft. If you are a Pulsaret v.1 user but you have not Coupon update discount or you have lost it, please
contact apeSoft ([email protected]).
See purchase a license and authorizes.
• background
The first official release of Density (2001), developed in Csound language and based on Eugenio Giordani ‘s
GSC4 (Granular Synthesis for Csound). GSC4 was the first patch for granular synthesis on Csound
implementing Barry Truax model.
Density can generate thousands of grains dynamically, I preferred this way, rather than a fixed number of
"voices" (oscillators). The overlapping factor grains, depends only on the actual CPU power. Thus you have
not limits in grains number for second (density).
Below an easy granulation model.
DensityGSC (Csound version) is still available for free at: www.alessandro-petrolati.it/densitygsc.html it
works on Windows XP but not in Windows Vista, it seems magically resurrected with Windows7.
DensityGSC is a discontinued product.
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New Pulsaret is completely rewritten in Max/Msp 5, available for both Macintosh and Windows, more stable,
flexible, improved audio quality, restyling GUI look (Graphic User Interface) with native effects Hv_pads,
FiltersEQ, Snapshots Sequencers improviser unit etc...
• GUI layout
The most important Pulsaret parameters are placed inside of only one window. Pulsaret GUI (Graphic User
Interface) is divided in three parts, WINDOWING MODULE (top left), MAIN MIXER (below) and GRANULAR
STREAMS (on the right). Many controls are accessible via pop-up menù. WINDOWING generate and edit
waves prototypes shapes, employed for envelope granulation, the MIXER, as a classical mixer, host
granular STREAMS (eight) gain faders, solo/mute switches, headphones monitor channel, native effects,
VST slots, I/O routing signals, MASTER fader (∑) and master/multichannel recorders.
Tools bar on the top, sets all Pulsaret preferences and open many Pulsaret functions like DSP settings, MIDI
I/O, Master/Multichannel Recorders HV_pads etc... Tools bar show graphically (icons) the menùs bar items,
tools bar it will be explained in the course of this handbook.
At the bottom of window, we have the status bar showing some informations like DSP settings, CPU %
usage; on the right - sound file informations and grains overlapping factor - for the granular stream selected.
You can select one of eight granular streams, clicking on the “stream selection tab”, also you can use
keyboard keys (shift+1,2...8) in case of shortcuts mode enabled.
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The Hint clue window (toolbar or menù bars), explains Pulsaret GUI widget, when the mouse is over it.
Pulsaret it’s easy to learn, thought for live performances. The main functions will soon be assimilated and
memorized, so, you can close clue hint window definitively, freeing the GUI from annoying hint.
• DSP settings
The engine DSP (Digital Signal Processing) can be activated clicking DSP toolbar (loudspeaker icon) or
loudspeaker icon in the MASTER channel. When DSP is turned on, both became orange color.
The audio DSP parameters are displayed in the DSP Status window. You can open DSP settings from
application menù View/DSP Cmd+D (Mac) or Ctrl+D (Win) or through the toolbar (loudspeaker icon).
The DSP Status window is arranged as a group of menùs and checkboxes that set all DSP parameters.
Status window serves as a monitor for your current audio settings as well.
At the left of the DSP Status window is a pop-up menù for turning the audio on/off and a set of pop-up menùs
that let you select an audio driver and configure its input source and output destination.
The second pop-up menù allows you to view and select an audio driver.
A brief summary will suffice for now:
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None: This setting shuts off audio processing.
Core Audio: This is the default audio driver for Macintosh. It interfaces with the system's built-in Core
Audio system and can be used with the built-in audio of the computer, or, with the proper software
support, a third-party hardware interface, such as ASIO.
MME or DirectSound: (Windows only) On Windows, Pulsaret loads the MME driver by default. If you
have correctly installed external hardware and it also supports DirectSound, it should also appear as
an option on the pop-up menù.
ad_rewire: This driver supports a standard developed by Propellerhead Software that allows sound
generating applications (ReWire Devices) to send multiple channels of audio and midi to other
applications (ReWire Mixers) that process and output it. Selecting the ad_rewire driver enables
Pulsaret to function as a ReWire Device to route audio from Pulsaret into applications that support
ReWire (such as Live, Digital Performer or Cubase).
ASIO: (Windows only) If you have a third-party audio interface which supports ASIO (a crossplatform audio hardware standard developed by Steinberg), and it is installed correctly, it will be
found by the Pulsaret ASIO driver. You may have as many ASIO devices as you wish; they will all be
found by the driver and will appear in the Driver pull-down menù in the DSP Status Window
preceded by the word ASIO.
ad_nonreal: This driver enables Pulsaret to work in non real-time mode, allowing you to synthesize
and process audio without any real-time processor performance limitations. Real-time audio input
and output are disabled under this driver.
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Only one audio driver can be selected at any given time. Pulsaret saves the settings for each audio driver
separately and will recall the last used audio driver when you restart Pulsaret.
The next two pop-up menùs are active only when using the Core Audio driver on Macintosh or ASIO drivers.
When the Core Audio driver or either the MME or DirectSound drivers on Windows are selected, the pop-up
menùs allow you to change the audio input source. These settings can also be changed using the Audio
MIDI Setup application on Macintosh or the Sounds and Audio Devices Properties window (Start > Settings >
Control Panel > Sounds and Audio Devices) on Windows, but only with these menùs while Pulsaret is
running.
When ASIO is in use, the pop-up menùs allow you to set the clock source for your audio hardware and
whether or not to prioritize MIDI input and output over audio I/O.
The DSP Status Window lets you control the size of the blocks of samples (called signal vectors) that
Pulsaret uses. There are two vector sizes you can control.
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The I/O Vector Size (I/O stands for input/output) controls the number of samples that are transferred
to and from the audio interface at one time.
The Signal Vector Size sets the number of samples that are calculated by Pulsaret objects at one
time. This can be less than or equal to the I/O Vector Size, but not more. If the Signal Vector Size is
less than the I/O Vector Size, Pulsaret calculates two or more signal vectors in succession for each I/
O vector that needs to be calculated.
With an I/O vector size of 256, and a sampling rate of 44.1 kHz, Pulsaret calculates about 5.8
milliseconds of audio data at a time.
The I/O Vector Size may have an effect on latency and overall performance. A smaller vector size may
reduce the inherent delay between audio input and audio output, because Pulsaret has to perform
calculations for a smaller chunk of time. On the other hand, there is an additional computational burden each
time Pulsaret prepares to calculate another vector (the next chunk of audio), so it is easier over-all for the
processor to compute a larger vector. However, there is another side to this story. When Pulsaret calculates
a vector of audio, it does so in what is known as an interrupt. If Pulsaret is running on your computer,
whatever you happen to be doing (word processing, for example) is interrupted and an I/O vector's worth of
audio is calculated and played. Then the computer returns to its normally scheduled program. If the vector
size is large enough, the computer may get a bit behind and the audio output may start to click because the
processing took longer than the computer expected. Reducing the I/O Vector Size may solve this problem, or
it may not. On the other hand, if you try to generate too many interrupts, the computer will slow down trying
to process them (saving what you are doing and starting another task is hard work). Therefore, you'll typically
find the smaller I/O Vector Sizes consume a greater percentage of the computer's resources. Optimizing the
performance of any particular signal network when you are close to the limit of your CPU's capability is a
trial-and-error process. That's why Pulsaret provides you with a choice of vector sizes.
Technical Detail: Some audio interface cards do not provide a choice of I/O Vector Sizes. There are also
some ASIO drivers whose selection of I/O Vector Sizes may not conform to the multiple- of-a-power-of-2
limitation currently imposed by Pulsaret's ASIO support. In some cases, this limitation can be remedied by
using the ASIO driver at a different sampling rate.
Changing the vector sizes does not affect the actual quality of the audio itself, unlike changing the sampling
rate, which affects the high frequency response. Changing the signal vector size won't have any effect on
latency, and will have only a slight effect on overall performance (the larger the size, the more performance
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you can expect). A signal vector size of 1024, it is 23.22 milliseconds. The Signal Vector size in Pulsaret can
be set as low as 2 samples, and in most cases can go as high as the largest available I/O Vector Size for
your audio driver. However, if the I/O Vector Size is not a power of 2, the maximum signal vector size is the
largest power of 2 that divides evenly into the I/O vector size.
You can set the audio sampling rate with the Sampling Rate pop-up menù. For full-range audio, the
recommended sampling rate is 44.1 kHz. Using a lower rate will reduce the number of samples that Pulsaret
has to calculate, thus lightening your computer's burden, but it will also reduce the frequency range. If your
computer is struggling at 44.1 kHz, you should try a lower rate.
N.B. Every Pulsaret project save current Sampling Rate value, thus every project can have differents
Sampling Rates.
The Scheduler in Overdrive option enables you to turn Pulsaret’s Overdrive setting on and off from within the
DSP Status window. When Overdrive is enabled, the Pulsaret event scheduler runs at interrupt level. When
overdrive is not enabled, the event scheduler runs inside a lower-priority event handling loop that can be
interrupted by doing things like pulling down a menù. You can also enable and disable Overdrive using the
Options menù. Overdrive generally improves timing accuracy, but there may be exceptions, and some thirdparty software may not works properly when Overdrive is enabled.
The Scheduler in Audio Interrupt feature is available when Overdrive is enabled. It runs the Pulsaret event
scheduler immediately before processing a signal vector's worth of audio. Enabling Scheduler in Audio
Interrupt can greatly improve the timing of audio events that are triggered from control processes or external
MIDI input. However, the improvement in timing can be directly related to your choice of I/O Vector Size,
since this determines the interval at which events outside the scheduler (such as MIDI input and output)
affect Pulsaret. When the Signal Vector Size is 512, the scheduler will run every 512 samples. At 44.1 kHz,
this is every 11.61 milliseconds, which is just at the outer limits of timing acceptability. With smaller Signal
Vector Sizes (256, 128, 64), the timing will sound ‘tighter.’ Since you can change all of these parameters as
the music is playing, you can experiment to find acceptable combination of precision and performance.
If you are not doing anything where precise synchronization between the control and audio is important,
leave Scheduler in Audio Interrupt unchecked. You'll get a bit more overall CPU performance for signal
processing The next portion of the DSP Status helps you monitor your system's performance.
The CPU Limit option allows you to set a limit (expressed in terms of a percentage of your computer's CPU)
to how much signal processing Pulsaret is allowed to do. Pulsaret will not go above the set CPU limit for a
sustained period, allowing your computer to perform other tasks without Pulsaret locking them out. The
trade-off, however, is that you'll hear clicks in the audio output when the CPU goes over the specified limit.
Setting this value to either 0 or 100 will disable CPU limiting.
The number next to Signals Used shows the number of internal buffers that were needed by Pulsaret to
connect the signal objects used in the current signal network. The number of Function Calls gives an
approximate idea of how many calculations are being required for each sample of audio. Both of these fields
will update whenever you change the number of audio objects or how they are patched together.
Vector Optimization only applies to PowerPC computers. Vector optimization allows four samples to be
processed within the space of a single instruction. However, not all audio signal processing algorithms can
be optimized in this way (for example, recursive filter algorithms are substantially immune from vector
optimization). Pulsaret itself no longer uses vector optimization, but third-party audio objects may still use it.
In other words, unless you are using a vector-enabled third-party audio object on a PowerPC computer, this
setting will have no effect.
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Mixer
• faders and solo/mute
Vertical faders (1,2...8) receive stereo granular stream audio signal. Master (∑) channel (on the mixer right)
sum all stereo faders and output audio data on the physical hardware audio device.
The 'mute' and 'solo' buttons, labelled 'M' and 'S' respectively.
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Activating the mute button mutes the channel, i.e. the affected channel can't be heard in the final
mix.
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Activating the solo button mutes every other channel, i.e. only the affected channel can be heard in
the final mix.
N.B. When you select one or more channels in solo, all the others channels automatically became muted.
Toolbar, (left to right)
The first toggle (back arrow) is sliders mouse (mousing) behave (no knobs and other widgets but onli
vertical sliders) with the follow options: relative and absolute. In relative mode (default) keeps it relative
position when you click on it. Moving the mouse outputs higher or lower values in relation to that relative
position. In absolute mode, will automatically jump directly to the clicked location.
N.B. mousing acts in the Mixer and in the granular streams parameters (vertical sliders).
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With the second tool, meters (central), you can specifies whether the faders displays and outputs the signal
level pre-fader or post-fader;
Most right fade-time set solo/mute fade in/out time in milliseconds, when you activating solo/un-solo or
mute/un-muted, the signal amplitude is smoothed with a linear ramp.
• I/O mapping
The Pulsaret mixer is multichannel, real number of inputs ADC (Ana-logic to Digital Converter) and outputs
DAC (Digital to Ana-logic Converter) depending to the hardware audio device. Each channel can be routed
on the MASTER channel, on the MONITOR channel (logical) or directly on a Physical hardware channel. By
default all streams goes in the MASTER.
Each stream also is routed on logical channel with predefined stereo number channels. Starting from the first
stream mapped on 17/18, second 19/20.... eight 31/32; instead the MASTER channel send audio on 1/2
logical channels while MONITOR send audio to 3/4 logical channels
Output signal is routed in three different ways: ∑ (on the left) address signal on MASTER channel (default);
‘m’ routing signal on MONITOR channel, explained below and labelled ‘o’ routing the signal in physicals
channels used by your hardware device, when selected you will see lighting the below menùs, from where
you can chose physicals hardware output ports (red highlight).
In the above example only two channels (stereo) are displayed, the outputs channels numbers is depending
by hardware device.
Using Core Audio
Core Audio provides audio I/O to Mac applications from both the computer's built-in audio hardware as well
as any external audio hardware you may have.
If you have external audio hardware, it should come the drivers to interface with Core Audio. When these
drivers are installed and the hardware is present, Core Audio will include the external device as a Core Audio
choice in the Driver menu in the DSP Status window.
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The Sound part of the System Preferences application can be used to set basic sound settings for the
system, such as the Output volume, left/right balance, and sound output device, as well as the Input volume
and sound input device. You can also use the Audio MIDI Setup application (located in /Applications/Utilities)
for more detailed control of the sound I/O settings. Note that modifications you make to the Sound section of
the System Preferences application, such as changing the output volume or balance, are reflected in the
audio MIDI Setup (and vice versa). You can open the Audio MIDI Setup application by clicking on the Open
Audio Control Panel button in the lower left corner of the DSP Status Window.
The Audio part of the Audio MIDI Setup application shows Input settings on the left side, and Output settings
on the right.
The System Settings let you choose which audio device is used for system audio input and output, while the
Selected Audio Device menu allows you to control the various settings for the built-in and any external
hardware audio devices.
When using external audio devices, the Input Volume and Output Volume sliders can be used to set the
overall input and output volumes of the selected device (they are not available when using the built-in audio
controller). The Device Mute checkboxes allow you to mute the input and output devices, if applicable.
Play Through is available on PowerPC Macs only. Play Through checkbox just under the Input Volume
slider lets you choose whether or not the input device is 'monitored' directly through to the output. When
playthrough is enabled, the dry signal from the input source will play through to the output mixed in with any
processed signal you may be sending to the output in Pulsaret. Disabling playthrough will enable you to
control how much (if any) dry signal from the audio input is routed to the output. The Input Section allows you
to select the Input Source (for example Line or Mic input for the selected device) as well as the sampling rate
and bit depth in the Current Format pop-up menu. Similarly, the Output Section also allows you to select the
sampling rate and bit-depth in its Current Format pop-up menu. The available selections will vary, depending
on your audio hardware.
You can set the volume levels for the individual audio input and output channels, mute individual channels,
and/or select them for playthrough using the controls located below the Current Format menus. The lower
part of the window is used to display the current input and output settings.
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Using MME Audio and DirectSound on Windows
Three types of sound card drivers are supported in Windows: MME, DirectSound and ASIO. Your choice of
driver will have a significant impact on the performance and latency you will experience with Pulsaret.
The MME driver (ad_mme) is the default used for output of Windows system sounds, and are provided for
almost any sound card and built-in audio system. While compatibility with your hardware is almost
guaranteed, the poor latency values you get from an MME driver make this the least desirable option for
real-time media operation.
DirectSound drivers, built on Microsoft's DirectX technology, have become commonplace for most sound
cards, and provide much better latency and performance than MME drivers. Whenever possible, a
DirectSound driver (ad_directsound) should be used in preference to an MME driver. Occasionally, (and
especially in the case of motherboard-based audio systems) you will find the DirectSound driver performs
more poorly than the MME driver. This can happen when a hardware-specific DirectSound driver is not
available, and the system is emulating DirectSound while using the MME driver. In these cases, it is best to
use MME directly, or find an ASIO driver for your system.
The best performance and lowest latency will typically be achieved using ASIO drivers. The ASIO standard,
developed by Steinberg and supported by many media-oriented sound cards, is optimized for very low
latency and high performance. As with the DirectSound driver, you need to verify that performance is actually
better than other options; occasionally, an ASIO driver will be a simple ‘wrapper’ around the MME or
DirectSound driver, and will perform more poorly than expected.
Using MME and DirectSound Drivers with Pulsaret on Windows
On Windows, Pulsaret loads the MME driver by default. If you have correctly installed external hardware, it
should support playback and recording with the MME driver and the Direct Sound driver in the Driver Menu
of the DSP Status Window.
If an audio device only supports MME or DirectSound, the Windows OS does an automatic mapping of one
to the other. Since many audio devices initially did not support DirectSound, Microsoft emulated DirectSound
with a layer that bridged from DirectSound to MME. Currently, there is greater support for native DirectSound
drivers, and sometimes when you use MME drivers Windows is actually running a layer to convert from MME
to DirectSound.
Note: Some devices such as the Digidesign mBox only support the ASIO driver standard. In such cases, you
will need to select the proper ASIO driver in the DSP Status Window. See the section ‘Using ASIO Drivers on
Windows’ for more information.
You can make overall changes to the basic operation of your default audio driver by accessing the Sounds
and Audio Devices Properties window (Start > Settings > Control Panel > Sounds and Audio Devices). Here
you can select Audio devices, and create settings for balance and output volume.
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Pulsaret supports the use of different input and output devices with MME and DirectSound drivers. Use the
DSP Status Window to choose input and output devices.
Input and Output Devices
When using MME or Directsound drivers, you may choose input and output devices from the pull-down
menus in the DSP Status window, which will be automatically populated with the drivers for your audio
hardware. When using the MME and Directsound drivers, it is possible to use different audio devices for
input and output simultaneously. However, this is not recommended or supported and unless there is some
external (from Pulsaret) provision for synchronizing the devices dropouts will likely occur over time.
Thread Priority and Latency Settings
Both MME and Directsound drivers include settings for Thread Priority and Latency. These are both set by
default to settings which we hope will work on your computer in the majority of situations. However, you may
find that when you are working with a patch that you have problems which you may be able to resolve by
changing some of these settings. If your audio is crackling or there are glitches in it, you may want to try
increasing the latency setting. This has the disadvantage of making your audio feel less responsive in real
time, but it will allow the audio driver more time to work on the extra audio demands you have placed on it.
If your system is slow in other areas -- such as screen redrawing or general timing accuracy -- you may wish
to decrease the thread priority of the audio driver. This will give your other tasks more room to get done, but
may also result in you needing to increase latency in order to give your audio driver room to breathe at the
new lower priority.
Timing between the control scheduler and Pulsaret is best when the I/O vector size is on the order of 1ms.
We recommend setting the IO vector size to 128 samples. Having a setting of the latency separate from the
I/O vector size allows this to work without audio glitches on most hardware.
Using ReWire with Pulsaret
The ad_rewire driver allows you to use Pulsaret as a ReWire Device, where Pulsaret audio will be routed
into a ReWire Mixer application such as Cubase. Both Pulsaret and the mixer application must be running at
the same time in order to take advantage of ReWire's services. The mixer application should be also
compatible with ReWire 2 or later for best results.
When the ad_rewire driver is selected, audio from Pulsaret can be routed to any of 16 inter- application
ReWire channels which will appear as inputs in ReWire mixer host applications. The first time ad_rewire is
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selected it will register itself with the ReWire system. Subsequent launches of ReWire Mixer applications will
then offer Pulsaret as a ReWire device.
For example, after the Pulsaret ReWire Device is registered, Cubase will have a Pulsaret menu item in the
Devices menu. When you choose it you will see a list of the audio outputs from Pulsaret. They will default to
the off state. Click on any of the buttons to activate that channel. Once the channel is activated it will show
up in the Cubase Track Mixer.
Using ASIO on Windows
Selecting an ASIO driver from the DSP Status window allows Pulsaret to talk directly to an audio interface.
To use ASIO soundcards your device needs to be correctly installed and connected; The Pulsaret ASIO
driver will find it at startup.
All correctly installed ASIO devices should be available to you for selection in the DSP Status window.
However, Pulsaret does not check to see if the relevant audio interface hardware is installed correctly on
your system until you explicitly switch to the ASIO driver for that interface card. If an ASIO driver fails to load
when you try to use it, the menus in the rest of the DSP status window will blank out. Switching to the MME
and/or DirectSound driver will re-enable Pulsaret audio.
The Clock Source pop-up menu lets you to set the clock source for your audio hardware. Some ASIO drivers
do not support an external clock; if this is the case there will only be one option in the menu, typically labeled
Internal.
The Prioritize MIDI pop-up menu allows you to set the clock source for your audio hardware and whether or
not to prioritize MIDI input and output over audio I/O.
Many ASIO drivers have other settings you can edit in a separate window. Click the Open ASIO Control
Panel button at the bottom of the DSP Status window to access these settings. If your interface card has a
control panel in its ASIO driver, the documentation for the interface should cover its operation.
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• monitoring
Activating the monitor button (labelled ‘m’) on the channel/s, the affected channel/s can't be heard in the final
mix, for instance you can hear sound in headphones.
The MONITOR it uses 3 and 4 logical channels, (i.e. Pulsaret MONITOR channels outputs).
In the example only stream8 send signal to MONITOR channel, clicking on the headphones icon you will
open Monitor window containing gain fader and output mapping menùs, now you must maps physical
channels (6/7 in the example). When one or more channels send signal to monitor channel, you will see
signal in the little meters below headphone icon.
N.B.
For monitoring signals you need a multichannel audio hardware device (four or more outputs channels).
• on/off stream trigger
Above Mixer channels we have streams triggers (on/off toggles). The same thing is achievable with on/off
toggle placed on the granular stream GUI, (see Granular Streams for more details).
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On/off buttons at the top of the master channel are shortcuts to turn on/off all granular strems together. They
work regardless of the of trigger mode selected (see below).
N.B. If DSP engine is off they will be disabled ignoring mouse clicks.
There are many ways to trigger (i.e. turn-on/off) granular streams:
• mouse click on/off button in the mixer or in the streams GUI;
• by sending a MIDI CC (Control Change);
• pressing spacebar keyboard, according to selected mode;
In order to use spacebar (computer keyboard) for triggering granular streams, you must enable toolbar
shortcuts toggle.
(from the left) enable/disable keyboard shortcuts, if enabled triggers menù mode will
be actived:
•none no spacebar trigger; current trigger only selected stream;
•subscribed streams (switch button in the mixer);
•all on/off eight streams together.
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When subscribe is selected, all eight buttons (little
orange) for subscription will be enabled. These are
located close to the on/off toggles.
Selecting one or more, you will be able to trigger them
with spacebar.
Shortcuts enable also following keys:
•
•
•
•
Cmd (Mac) or Ctrl (Win) + 1, 2, … 8 to switch among granular streams;
escape to hide chromatic keyboard in the streams GUI (if shown) and/or close dialogs messages box;
enter key to accept;
WavePad: select(F1); loop(F2); move(F3); draw(F4).
(see Granular Streams for more details).
• native effects
Every stream channels have own native effects, the four colored buttons display the window effect.
19
Topdown we have:
• snapshots-sequencer (labelled s);
• Filter multiband Equalizer (labelled f);
• Reverberation module (labelled r);
• Dynamic compressor (labelled c);
audio chain is processed as follow:
stream signal >>> sequencer >>> filterEQ >>> reverber >>> compressor >>> VST_slots
See snapshots sequencer improviser unit, FilterEQ and VST_slots for more explanations.
Reverberation module, a plate reverb in the style of Griesinger by Randy Jones.
20
Parameters (from left to right):
• dry/wet balance (%)
• T60 reverberation time (0÷∞)
• high frequency damping (Hz)
• room size (filters coeff.)
• diffusion (filters coeff.)
N.B. the last two right knobs (brown colored) are general settings reverber parameters, you should not move
it during performance, anyway values are mouse-filtered, their values pass only when the mouse button is
up.
21
Dynamic compressor is a dual-band fully-featured signal compressor with limiting, gating, side-chain.
on/off bypass dynamic compressor;
enable/disable meters, these values describe the current state of various
internal gain levels of the
compressor:
• compressor gain (left, right);
• noise gate gain (left, right);
• limiter gain (left, right)
The last two right menùs on the right sets
• limiter response mode: punchy or smooth. Punchy response yields extremely short attack and release
times, useful for transparent limiting, or to create loudness. However, if over-used, intermodulation
distortion may result. Smooth response uses longer attack and release times. The result is still a fast lookahead limiter, but with less intermodulation distortion and less punch.
• the most right chose one of build presets for the dynamic processor: Guitar, Bass, Vocal, Drums and
Program Material - An attempt at smooth "gain riding" of mixed program material as well as can be done
with a non-multi-band processor.
threshold sets the compressor threshold (in dB below full scale). This is the main compression threshold.
Any signal above the threshold will be reduced, and any signal below the threshold will be amplified,
according to the range and ratio parameters;
ratio sets the numerator of the compressor gain reduction ratio, from 1:1 to Infinite:1;
attack sets the rate at which the compressor is engaged when the signal level exceeds the Threshold. The
value range is 0-150 on a logarithmic scale, with larger values indicating faster attack;
release sets the rate at which the compressor releases its gain adjustment when the signal level no longer
exceeds the Threshold. The value range is 0-150 on a logarithmic scale, with larger values indicating faster
release. This rate can be modified by the release gate and freeze thresholds.
• VST slots
22
The last effect of the audio chain is a VST 1 slot.
There are nine VST slots in Pulsaret, for the eight granular streams and the
MASTER channel. Load and manage VST plugin is easy. Clicking on menù
(double arrows), loadVST item allows you to choose a VST plugin file (.vst
on Mac or .dll on Windows); open, show the VST effect window; empty
cleans current VST slot which means no plug-in (default); bypass enable or
disable VST effect; write/read export/import VST bank file; the lowest item
show the path and name of VST loaded. Clicking on pathname you will
also open VST GUI (as the open item).
All VST settings can be exported inside of Pulsaret project folder.
Some VST are not fully compatible with Pulsaret therefore you may not
successfully export their contents.
• Recorder
You can capture all Pulsaret audio signals in two different ways: "master rec" and/or "multichannel rec". From
menùbar, toolbar or ‘R’ button (on the MSTER channel) you can open Recorder windows.
Before to start recording, you need toggle on ARM recorder unit. You can change output file Type
(quantization bit deep) and Sampling Rate. Now you will see stereo signal through the V-meters. You are
ready to record (click on Record toggle).
• Master recorder have only an horizontal fader witch you can controls the signal gain (post-master).
Multichannel recorder show simply the 16 V-meters for the granular streams;
• Sampling Rate (default 44100) for the output file is taken by the current DSP setting. If Pulsaret SR is
different from File out SR, the file output it will be resampled;
• File Type (quantization bit deep) default 16 bits;
folder/file targets switch: in folder mode (default) a new audio
file is created each time on starts a recording. The file name is
generated by using the current date and time in the machine, in
order to avoid name conflicts, and to reflect the order of
recording. The default location where the file is recorded is the
Pulsaret Application folder. You can change location by clicking into the "choose dir" button, and select a
1
Steinberg's Virtual Studio Technology (VST) is an interface for integrating software audio synthesizer and effect plugins with audio
editors and hard-disk recording systems. VST and similar technologies use Digital Signal Processing to simulate traditional recording
studio hardware with software. Thousands of plugins exist, both commercial and freeware, and VST is supported by a large number of
audio applications. The technology can be licensed from its creator, Steinberg. (Wikipedia)
23
specific place. If one wishes to set a different name and/or location for each new file, it is possible to use the
"chose file" button. Once a file is created, each new recording will be written into it. When one doesn't want
to overwrite its previous content, one should create a new file again.
You can switch anytime between folder/file and you can also use pop-up menùs where is displayed file/folder
path. The mode which is currently active has a red color.
Multichannel recorder write a multichannel audio file containing all the eight granular streams signals (i.e.
stereo channels 2 x 8 streams = 16 channels).
Unlike the Master recorder, in Multichannel you cannot rescale channels record gains.
N.B. master/multichannel recorders, they receive audio signals before to being sent to a physical or logical
channels, so even if the output is disabled (off) or muted ‘M’, they will continue to receive and recording the
signal normally.
N.B.
• You can recording with master and channels recorders simultaneously;
• Record start/stop it will be not saved in the Pulsaret project;
• If Master or Multichannel Recorders are at work, you will see ‘R’ button (on the Mixer Master channel) to
flash.
24
Filter Equalizer
• general
One of the most important and powerful Pulsaret feature is the multi-band filter equalizer. You can
dynamically employ up to 24 two-pole filters (i.e. biquad), often referred to as "second order sections".
In the Graphical filter editor you can set higher-level parameters such as frequency, amplitude and
resonance (Q or S) through the mouse or selecting a filter band and using fine sliders.
Tools bar is divided in three rows. Follow explanation top-down and left-right:
on/off switch to enable/bypass filter;
peacknotch (default) menù sets a filter kind: 0 - display only 1 - lowpass 2 - highpass 3 - bandpass 4 bandstop 5 - peaknotch 6 - lowshelf 7 - highshelf 8 - resonant (see filter table at the end of this section);
numbox (showing 4 in the above image) sets the number of cascaded biquad filters displayed in the
filtergraph, the range is between 1 and 24 (default 4).
analog toggles the analog filter prototype parameter for the bandpass, and peaknotch filters. Unlike the
standard digital versions, these "imitation analog" filters do not have a notch at the Nyquist frequency, and
thus imitate the response of a analog filter. The imitation analog filters are slightly more computationally
expensive, so this option disabled by default;
gain toggles the gain parameter for the lowpass, highpass, bandpass, and bandstop filters. The traditional
definitions of these filters have a fixed gain of 1.0, but by gain-enabling them, their amplitude response can
be scaled without the additional use of a signal multiply at the filters output.
log freq, log amp and ph spect sets editor view range:
• log/linear frequency display;
• log/linear amplitude display;
• view phase spectrum -pi to pi;
• show parameters when mousing over the editor;
• resets all filters gains;
25
select a region of the spectrum for frequency zoom, Shift-clicking to extends the range; Command-clicking
(Mac) or Control-double-clicking (Win) and dragging shifts the current range up or down. Option-clicking
(Mac) or Alt-clicking (Win) and dragging up or down expands or shrinks the range. Command-double-clicking
(Mac) or Control-double-clicking (Win) selects the entire range.
26
Tab numbers (in the example is selected 7), select a detail filter band.
• cutoff/gain
N.B. filters band could be not progressive in frequency, therefore you could have, for instance, the second
cut-off frequency upper of the follows bands and vice versa.
Three sliders below sets respectively:
• cutoff (center freq) in Hz of the currently selected filter;
• gain in db of the currently selected filter;
• Q (resonance) or S (slope - used for the shelving filters) of the currently selected filter;
you can also set the cut-off frequency, opening the musical keyboard and clicking on the keys. Keyboard
control also the gain of selected band filter, mouse clicking in different heights on each key.
If shortcuts is enabled, pressing escape key you will hide keyboard. If toggled the most right keyboard (most
right key) switch, the keyboard will be always in foreground (disabled auto exit when click on a key).
The lowest sections is considered improvisation unit. You can experiment settings filters band with one of the
COMPILERS (red frame); manage randomly or together gains/Q (purple frame) and make manually an
equalization for each band (blu frame).
dry/weet balance between filtered/un-filtered signal, you can also set dry/wet fade-time in ms (25 ms
default);
• compilers
...inspired to Eugenio Giordani Stria Multilevel Interactive Sound Synthesizer.
They increase complexity and coherence between the filters band space. You can create your pitch space by
the way of four different criteria of construction by determining a pitches grid. The frequency range is
depending by fundamental frequency and numbers of band filters. When a frequency exceed Nyquist1
frequency (i.e. SR/2) the filter band generation is inhibited.
• Fibonacci COMPILER
1 The
Nyquist frequency, named after the Swedish-American engineer Harry Nyquist or the Nyquist–Shannon sampling theorem, is
half the sampling frequency of a discrete signal processing system. It is sometimes known as the folding frequency of a sampling
system. The sampling theorem shows that aliasing can be avoided if the Nyquist frequency is greater than the bandwidth, or maximum
component frequency, of the signal being sampled. (Wikipedia)
27
Fibonacci 1 COMPILER: the ratio of two consecutive Fibonacci numbers converges on the Golden Mean
Ratio (approximately 1:1.618) as its limit.
The computer music composition Stria by John Chowning 2 was based in almost every structural aspect on
this ratio. Fibonacci compiler uses this sequence of numbers for compile the grid pitches.
You can set two parameters for the purpose to create a certain Fibonacci pitch grid. You have to specify:
• base frequency in Hz i.e. fundamental frequency (left slider 35.1 Hz) for the Fibonacci spectrum;
• first member of the series (value of 2 in the example).
To complete the operation and generate the pitch grid you must press the compile button.
When auto is enabled, changing value of one of parameters, (for each compilers methods) grid pitch it will
be automatically compiled.
• Geometric COMPILER
Geometric COMPILER create “geometric pitch space” based on a ratio number. For example if you specify
1.5 (i.e. 3/2), you will create a pitch space grid based on the natural interval of 5th. Since you can choose
any relationship, you can virtually have infinite pitch spaces.
Interval ratio defines a coefficient of multiplication, this serves to build up the whole pitch grid while the
fundamental frequency is the starting point from which grid take origin.
• base frequency in Hz (fundamental frequency) for the geometric spectrum;
• ratio for the geometric spectrum ( i.e. 1.6 also represented by the ratio 8/5)
In the below example, a geometric pitch grid is based on the interval 1.6 and fundamental frequency 55 Hz.
The grid will consist of a total of 13 bands starting from 55 Hz up to 15.481,1279 Hz as sketched in the next
table:
1
Leonardo Pisano Bigollo (c. 1170 – c. 1250)also known as Leonardo of Pisa, Leonardo Pisano, Leonardo Bonacci, Leonardo
Fibonacci, or, most commonly, simply Fibonacci, was an Italian mathematician, considered by some "the most talented western
mathematician of the Middle Ages. (Wikipedia)
2
John M. Chowning is known for having discovered the FM synthesis algorithm in 1967. In FM (frequency modulation) synthesis, both
the carrier frequency and the modulation frequency are within the audio band. In essence, the amplitude and frequency of one
waveform modulates the frequency of another waveform producing a resultant waveform that can be periodic or non-periodic depending
upon the ratio of the two frequencies. (Wikipedia)
28
You can obtain the same result using the next recursive formula:
fi = f(i‐1) * iterval_ratio
where fi is the i-th frequency (i.e. f2 = f1 * 1.6 = 55.00 * 1.6 = 88.00).
• Harmonic COMPILER
Harmonic COMPILER creates a bands grid according to the natural series of integer numbers (1,2,3...etc),
starting from a fundamental frequency. There are two different ways to do it:
• base frequency in Hz (fundamental frequency) for the harmonic spectrum;
• INHARM factor for spectral expansion/compression. You can stretched or compressed harmonic spectra
putting the INHARM parameter at a non zero value;
In addition you can create stretched or compressed harmonic spectra putting the inharm parameter at a non
zero value. Stretched spectrum is obtained from values greater than zero, while for compressed spectrum
you have to select values less than zero.
The stretched or compressed spectra are created using the following formula:
where:
Fn = n-th frequency
F0 = fundamental frequency
n= index of Fn frequency
exp = factor of expansion/compression (inharm)
The inharm factor can vary from -0.2 +0.2 When inharm = 0 the generated frequency grid is pure harmonic.
• Scalar COMPILER
29
Scalar COMPILER method distributes linearly all bands from freq min to freq max range.
• minimum frequency for the scalar spectrum:
• maximum frequency for the scalar spectrum;
Alternatively you can impose the gain (top slider 19 dB in the example) and Q (below) at all filter bands in
the spectrum. Also randomize all values.
An hidden function in the equalizer multi-slider, is Equalizer gain (0 ÷ 56 dB).
Mouse scrolling up/down on the value, you can change filters gains. This value must be considered bi-polar,
so in the above example 38 dB it means -38 +38 equalizer range.
• transitions subscribe
You can subscribe/unsubscribe the Filters for snapshots transitions. The transitions on the Filters parameters
work fine when you make interpolation between 4 (or more) stored snapshots. Default is un-subscribed. For
instance you can take a snapshot of current filters (bands) configuration and move them toward a second
snapshot making a linear interpolation between all filters bands.
30
Granular Streams
• parameters
In this section we will examine the Granular Synthesis parameters. Please refer to web for more
explanations about granular synthesis.
You can interact with Pulsaret parameters on the GUI (Graphic User Interface) in the follows ways:
1) Click and drag (scroll) on vertical sliders (or dials) to change the value, (see mousing mode for more
details);
2) Holding down the Command key (Macintosh) or the Control key (Windows) while mouse scroll, for precise
value control;
3) Select a widget clicking on the name and use keyboard up/down arrows;
4) Drag up/down the display value (the number box below each parameter);
5) Click on the number box value, enter a numeric value and press enter key;
6) Send a MIDI CC (Control Change), (see MIDI/OSC I/O chapter in this manual);
7) Send an OSC (Open Sound Controller) message (see MIDI/OSC I/O in this manual);
The granulation engine is updated every new generated grain, the frequency of grains scattering depending
from Pulsaret parameter, expressed in Hz (grains for seconds).
The vertical sliders provides a deterministic values, while the dials (knobs) provides random values.
Summing the two values (deterministic-random) will be produced a value that is passed to the granulator.
In the follow example, the pitch value (deterministic) is rescaled with rnd_pitch (random):
pitch_result = pitch + ( birnd(rnd_pitch * pitch/2) )
Where birand is bipolar random generator. The random range is expressed in %, therefore 0 % it means
only deterministic value, 100 % pitch_result can vary in the range:
pitch_result = pitch ± pitch/2
31
In the example rnd_pitch (green frame) starting from 80%, is reduced to 5 % in the middle.
With a great rnd_pitch, each grain can have an high random pitch mask range. Carrying rnd_pitch toward
lowest values, the random pitch mask is reduced until 0 % which means no random pitch grain scattering.
length slider = grain duration in milliseconds;
rnd_length dial = random range deviation, in % of length;
density slider = grain for seconds in Hz;
rnd_density dial = random range deviation, in % of density;
cps = grain frequency in Hz;
freeze button = (small button in the middle of scanning control) switch between the value 0 (freezes) and the
last control value;
rnd_cps = random range deviation, in % (12 semitone);
glisson = glissando semitones inside of the grain;
rnd_glisson = randomly moves semitones glisson (-/+ 12 semitones);
morph = morphing between two waveshapes (left=wave_1, right=wave_2, middle=mix 50%);
rnd_morph = morphing randomply between two waveshapes, in %;
volume slider = grains amplitude in decibel;
pan dial = left/right grains distributions on the stereo front (only pan = manually);
distance slider = stereo field width;
panningmode tab = select between: pan (manually), jittering (random), lfo (low frequency modulation);
beat/Hz toggle (only lfo) = toggle between beat/Hz; sets the panning duration (left right azimuth) expressed
in note-ratios or frequency (Hz). Beat mode is synced with the global transport time, you can change global
time (BPM 1) from the global transport.
beat menù (only lfo) = each item switch represents the Relative-Tempo expressed in note-ratios; you can
change BPM (Beat per minute) from the global transport. N.B. you need activate the global transport master
clock.
panning rotate frequency slider = when selected lfo, set the panning azimuth in Hz, you can chose a
shape (sine) among those available (see Windowing section for more details).
1 Beats per minute (BPM) is a unit typically used as a measure of tempo in music.
The BPM tempo of a piece of music is conventionally shown in its score as a metronome mark, as illustrated to the right. This indicates
that there should be 120 crotchet beats (quarter notes) per minute. In simple time signatures it is conventional to show the tempo in
terms of the note duration on the bottom. So a 4/4 would show a crotchet (or quarter note), as above, while a 2/2 would show a minim
(or half note). In compound time signatures the beat consists of three note durations (so there are 3 quavers (eighth notes) per beat in a
6/8 time signature), so a dotted form of the next note duration up is used. The most common compound signatures: 6/8, 9/8, and 12/8,
therefore use a dotted crotchet (dotted quarter note) to indicate their BPM. (Wikipedia)
32
All the granular parameters have a prefixed values range. Some of them have an additional feature, you can
divide or multiply the parameters output in order to extend range.
The rescaled value it will be passed to granulator engine, but on display you will see old prefixed widget
range.
• settings
Streams setup:
-on/off granular stream works just like the play/stop buttons on the mixer,
you can either use the space bar (see pop-up menù spacebar stream
trigger).
-auto/manual enable disable auto grain scheduling. If disabled you can
trigger manually a new grain, default value is auto then grains trainlet
depending by density/rnd_density parameters.
- envelope/wave interpolation mode N.B. low interp is faster than high, but has more interpolation artifacts
and cannot play buffer~ of arbitrary length.
• grain envelope
From the windowing envelope pop-up menù, you can choose a shape for granulation
or loading until 12 sound files. Each stream can have a different envelope, the default
shape is “draw”. when you load an audiofile (or mp3) in windowing, you will see file
name in the streams envelope select popup menùs.
see Windowing in this handbook.
• wave select
33
Pulsaret provide a powerfull way to mix two waveshapes, every grain it will be initialized with own % mix of
twi shapes. You can reach edge shapes by selecting differents wave shapes from menùs
Windowing
• general
The windowing module generate some classics envelopes and prototypes used for smoothing the grain
amplitude. You can deforming some envelopes shapes like gauss, curve or additive, draw a new shape
freehand or through draw prototype. Also you can load an audio file from disk (until 12 audio files, Aiff Wav
or Mp3 supported). Almost all Windowing WavePad functions are identical to Granular Streams WavePad.
See WavePad, WavePad display, WavePad Snap, WavePad mouse for more details.
Unlike Granular Streams WavePad, the Windowing WavePad buffer management have two additional
functions:
• reset shape
• reset all shapes
When you select from menù the first of them, current envelope or shape will be restored at the original
shape.
When you select reset all shapes, all prototypes and envelopes they will be restored at the original shape.
resetshape and resetallshapes, they work on the sixteen envelopes/prototypes restoring originals shape.
While clear and clearall they work either on the sixteen envelopes/prototypes either on the twelve sound
files slots, writing 0 in the all buffers.
34
Windowing have an important parameter that allow you to chose
a size in samples for prototypes or envelope shape. You can also
resize buffer dynamically, entering a new value expressed in
samples (512 default).
N.B. when you resize a shape, you will reset original default
shape.
• shape select
There are 16 pre-generated shapes:
1.
2.
3.
4.
5.
6.
7.
8.
9.
10.
11.
12.
13.
14.
15.
16.
rectangle,
blackman,
curve,
gauss,
hamming,
hanning,
draw,
sine,
cosine,
triangle,
saw,
square,
sigmoid,
sin(x)/x,
random,
additive
Some of them have additional controls parameters (bold). Selecting one of them, their additional parameters
will be shown.
curve show three box-number, on the top (top right 150 ms) we have deformation update rate in
milliseconds. Since these parameters work in real time, you can set a speed limit time to avoid cpu
overhead. The other two parameters (1.00 and -0.72) sets middle/border curve deformation.
35
Linke curve, gauss show on the top the speed limit update rate and only a “Gaussian standard deviation”
parameter.
• draw
In Pulsaret, the grain envelope shape default is draw (bold italic), when selected appear yellow button (draw/
done). This allows you to superimpose at the WavePad a function break-points where you can track the
envelope segments. You can add new break-points by clicking on the function superimposed pad or shift +
click on a specific break-point to remove it.
Two defaults shapes are available, adsr (Attack Decay Sustain Release) and a trapezoid (default grain
envelope). You can switch between unipolar/bipolar automatically when you change view perspective in
windowing WavePad, (see Granular Streams WavePad display for more explanations).
• additive
additive generate a composite waveform through weighted sums of sinusoids. It works similarly to Csound
Gen19. Each slider in the additive pad, controls a specific partial amplitude. You can change the total
36
harmonics number for the spectrum (top right numbox 18), clear all amplitude contributes of spectrum
(bringing to 0). Toggle on auto, you will enable amplitude auto-rescale, thus the resulting shape it will be
always normalized to one. N.B. phase and offset are normalized (0 ÷ 1). The two below num-box of gen
button, sets the fine amplitude of a specific partial. You need selects a partial number (1 in the example),
enter the amplitude value (0.5 in the example) at last click on gen to apply change.
• audio slot
In the shape select menù, after the sixteen envelopes/prototypes you will see twelve empty slots named:
null. You can employ them to load until twelve audio files form disk, just like Granular streams WavePad.
See for further details. If you try to load, by dragging or from the menù, a sound file on a shape envelope or
prototype, you will see a message box:
while if you drag on the Windowing WavePad a folder contains audio files, the slots they will be automatically
filled. See WavePad, WavePad display, WavePad Snap, WavePad mouse for more details.
• WavePad
On the right side of the WavePad, you can choose the type of action: select, loop, move and draw.
The tools select (from top to bottom) and loop allows you to select a buffer length
move (hand) allows you to select a part (zoom) of prototypes or loaded audio file.
• WavePad display
37
Clicking Normalize to scale the sample values in the WavePad buffer (0 dB), so that the highest peak
matches the value of 1.
Clicking Hzoom to reset horizontal zoom (entire time display of file);
Clicking on Vzoom you resets vertical amplitude zoom (default 50 %). Mouse scrolling on right number (50
%) to set an amplitude zoom amount. N.B. unlike Normalize, Vzoom is not a destructive amplitude rescaling
but only a graphics rescale.
In the image below, the last right menù (bipolar) switch to unipolar/bipolar view mode, unipolar mode shows
values between 0 ÷ 1 while in bipolar shows values between -1 ÷1;
• WavePad Snap
You can set the WavePad snap mode selection range:
bipolar switch wiev mode unipolar/bipolar;
Snap causes the start and end points of the selection to automatically move to specific points in the buffer,
defined by the snap mode:
•nothing disables snap to allow free selection. This is the default;
•grid specifies that the selection start and end points (grain length) should snap to
the vertical grid lines;
•zerocrossing instead of snapping the selection to a uniform grid, this mode
searches for zero-crossings of the buffer data;
The seconds menù from the top (millisec...), sets the unit of time measurement used by the display:
•milliseconds sets the display unit to milliseconds;
•samples causes time values to be shown as sample positions in the target buffer. The
first sample is numbered 0, unless the display has been shifted by the offset message;
•phase causes time to be displayed according to phase within the buffer, normalized so
that the 0 refers to the first sample, and 1 refers to the last; N.B. grid values for phase,
must be normalized (0. ÷ 1.), (see later);
38
• bpm specifies beats per minute as the time reference unit, relative to a master tempo and number of beats
per bar, both of which you can set with the bpm message.
The numbox (1.00) specifies spacing of the grid lines for current
unit of time measurement used by the display. N.B. grid values for
phase, must be normalized (0. ÷ 1.)
• WavePad management
The top-left pop-up menù (Canto.wav ), select one of the twelve waveforms to display on the pad. Current
"granular stream” will granulate selected waveform. You can load/replace a sound file from menù or dragging
file (or folder files) on the WavePad. The menù “null” item, it will be replaced with the file name.
load/replace soundfile = open a file browser to load an audio file on
the selected slot (aiff, wave mp3 supported). Also you can drop a
valid file on the wave pad, if it’s unsupported, a message will be
shown;
load/replace folder (soundfiles populate) = open a file browser
to chose a folder to fill sound buffers audio files (aiff, wave mp3
supported). Also you can drop on the WavePad a sound files
folder, the unsupported files will be not loaded and a message will
be shown;
save buffer to file = export the contents of the buffer as Wav or
Aif audio file;
clear buffer = erases the contents of buffer;
clear all buffers = erases the contents of all twelve buffers;
trim selection = will trim the audio data to the current selection. It
resizes the buffer to the selection length, copies the selected
samples into it, and displays the result at default settings. The buffer is erased, except for the selected range.
This is a "destructive edit," and cannot be undone;
undo last selection = it causes the selection start and end points to revert to their immediately previous
values. This is helpful when you are making fine editing adjustments with the mouse and accidentally click in
the wrong place, or otherwise cause the selection to change unintentionally. Repeated undo commands will
toggle between the last two selection states;
zoom out = resets the entire Hzoom (horizontal zoom) time display;
edit files infos = show a text-edit with information about loaded files;
open buffer window = opens the WavePad buffer window, or brings it to the front if it is already open. The
windows is resizable but not editable, you can scrolling the mouse over the buffer window;
normalize (1.00) = scale the sample values in the buffer so the highest peak matches the value given by the
argument. This can cause either amplification or attenuation of the audio, but in either case, every value is
scaled. When a sound file is loaded, default value is 0, it means original wave amplitude, value greater then
0 are saved and loaded in the project. N.B. Normalized value of 0 can be interpreted like:
1. original file amplitude if you have not change it before,
2. normalizing at 0 value (i.e. 0.001 amplitude gain, -60 db).
39
Snapshots
• store/recall
A snapshot is a photo of graphical interface (GUI parameters) in the current state. Each snapshots module
can store up to 100 snapshots. In Pulsaret there are ten modules "snapshots":
• main manage main mixer parameters;
• streams 1 ... 8 manage granular stream parameters;
• gsnap (below image) manage all Pulsaret parameters (main mixer and streams);
You can make transitions between two or more snapshots in more streams and/or main mixer
simultaneously. You can subscribe or unsubscribe determinate Pulsaret parameters.
N.B. gsnap transitions works on all Pulsaret parameters, therefore no local transitions should be at works.
This in order to avoid more transitions controlling the same parameter.
When you find an Interesting "sound" you can store new snapshot just shift + click on a button in the snappad, thus the preset button will light orange. You can recall a snapshot by simply clicking on the snap-pad
buttons; all the widgets subscribed (see later) will be restored at current snapshot value, immediately.
Alternatively a snap can be stored from pop-up menù, a message box will ask you to digit a snap number in
(see below for more explanations).
• micro pad
The micro-pad’s goal is to obtain intermediate values between four snapshots (interpolation). You can
configure four nodes (snapshots).
locked (lowest in the image): nodes: the mouse can only edit the nodes position and size. unlocked:
slider; the mouse can only changes the slider location.
refresh (highest in the image) enable/disable auto-recall snapshots, see below for more details.
• transitions
Transitions are a linear interpolation between two snapshots, with one time duration.
You can start a transition only from the menù recall item.
Is important to understand that the transition occurs from the current parameters positions, toward the
selected snapshot.
40
actual GUI widgets positions >>> (toward) selected snapshot (time duration)
You can change the transition-time (2.00 sec in above image). The transitions can occur simultaneously on
the streams and/or on the main. If auto-recall (see above) is enabled, selecting from the menu a snapshot
slot to start the transition.
• snapshots manage
-store new snapshot using the next empty preset slot;
-store snap number, enter a number or a list separated by space, to store;
-clear current selected snapshot;
-clear all snapshot, a message box will ask you to confirm;
-rename, open a window list where you can name or re-name your snapshots;
-renumber sort all snapshots stored into consecutive, beginning with 1;
-recall1 start a transition toward current snap number;
-pause/resume current transition (if occur);
-stop transition cancel current transition (if occur);
-default params, resets subscribed parameters at default value, only for the
current streams or main 2;
-defaults all resets subscribed parameters and general settings (not subscribed)
at default value, only for the current streams or main 3;
-save to disk current snapshots-bank;
-load from disk current snapshots- bank;
-client window, open clients list to subscribe, unsubscribe, setting interpolation
type and more;
-storage window open storage window displays any stored presets;
-open sequencer improviser unit (see Snapshots Sequencer);
-open HV_PADS (Hyper Vectorial Pads, see HV_Pad);
-speedlimit set update time limit for transitions and micropad nodes interpolations.
N.B. all the presets are saved in the project (see save/load project further), however, you can manage
individually save/load snap-bank, for example useful for exchanging presets between the streams.
• rename window
Opening rename window to fast renaming snapshots:
1
you can enable or disable auto-recall, if enabled selecting a snap from the menù, it will start the transition, if disabled it will be only
selected.
2
3
gsnaps will restore all Pulsaret subscribed parameters at default values (eight streams and main mixer).
gsnaps will restore all Pulsaret subscribed parameters and general settings (not subscribed) at default values (eight streams and main
mixer).
41
double clicks on the <(unnamed)> to edit text, then rename it.
You have an additional feature, scrubbing on the left column numbers, you will able to switch between
snapshots. Thus this secondary unit can be considered important when you experiment your sound or during
performance.
N.B.
Pulsaret store defaults values every time you open the application in the slot 99. Slot 99 contains either
default parameters value, either defaults all values (parameters and general settings).
Slot 100 is where Pulsaret store the last GUI state (this occur when you save/load a project).
• subscribe/un-subscribe transition clients
The two windows client and storage, are both non-interactive:
42
client window (accessible from the client window menù item, see above) shows the current subscriptions
widgets list (ever header name), priority, interpolation and data belonging to subscribed widgets.
A few types of interpolation can be changed:
• off: no interpolation;
• linear: Linear interpolation. Presets recalled will be interpolated using a standard linear algorithm.
• threshold: Threshold. Takes optional argument (float), which sets the threshold. Presets recalled will recall
data from the first preset specified when the fade amount is below the threshold, and will recall data from
the second preset specified when the fade amount is greater than or equal to the threshold. e.g. threshold:
'fade' < thresh = value a; fade >= thresh = value b;
• inverted threshold: Inverse threshold. Takes optional argument (float), which sets the threshold. Presets
recalled will recall data from the first preset specified when the fade amount is greater than or equal to the
threshold, and will recall data from the second preset specified when the fade amount is less than the
threshold. e.g. inverted thresh: 'fade' < thresh = value b; fade >= thresh = value a;
• exponential curve: Power curve. Takes an additional argument (float), which sets the exponent to which
the fade amount will be raised. Presets recalled will recall data between the two specified presets, along
the curve described. Power curves can be used to create faster or slower "attacks" and "decays" for the
fade envelope;
• table: Table-specified curve. Takes optional additional argument (see transition curve below), which
specifies the name of a table to use for curve lookup (tab1, tab2, tab3 or tab4). Presets recalled will recall
data between the two specified presets, along the curve described in the table. Tables are assumed to
contain values between 0 and 100, representing the new fade amount * 100. If the lookup fade amount
does not fall exactly onto a table-specified value, linear interpolation is used to determine the new fade
amount. In Pulsaret you have four draw functions to draw your curve, see transition curve below.
The storage window displays any stored presets. The active (recalled) preset is displayed in highlight
green. If any client value is changed, are displayed in italics. Eventually, both of these windows will be
configurable and editable, so that they can provide display and editing control for clients and storage sets.
N.B. When you open storage window, a message box will be displayed:
43
this is to avoid an overhead of CPU when displaying storage window.
• transition curve
All transition by defaults are linearly interpolation. You can superimpose your interpolation curve at one or
more subscribed widgets. Four draw functions are available, from toolbar or menù bar. Open one of them
and draw your curve. This feature is saved on Pulsaret project.
Draw a shape on the function window
In order to maps one or more subscribed clients with one of four functions draw, you need open clients
window and select table from pop-up menù (over interp header). Now you must digit the table name, that
can be: tab1, tab2, tab3 or tab4.
44
in the above example, we employ:
• tab1 like transition curve for density, rnd_density, rnd_length;
• tab2 like transition curve for scrubb;
• pan, pitch, rnd_pitch and scann are unsubscribed from the snapshots transitions system.
See client window above for more explanations.
45
Snapshots sequencer improviser unit
Snapshots sequencer rhythm improviser unit i.e. the extended name, the aim of this device is to control the
snapshot sequence88 by improvising with different rhythmic pulses. The sequencer shown on vertical the
snapshots numbers while in horizontal steps sequence.
• beats cycle
A metronome is synced to global transport and outputs a trigger on each beat. The beats are counted in
what we're defining as a a cycle of <n> beats (Cycle sub-patcher). Each time the cycle restarts, we're
choosing whether or not we want to change the current time interval. This is set by using a probability factor
(ChangeProb sub-patcher). If the program decides to change the time interval, a trig is sent.
• time intervals
Since eight time intervals can be selected. The choice is performed with a random function, according to a
weight probability for each possible interval - an interval with a weight of 2 will have twice as many chances
to be selected by the random procedure that an interval that has a weight of 1.
• step sequencer
When a given time interval is chosen, it is used as a step rate for driving the step-sequencer. The stepsequencer is synced to global transport, so the steps are played in accordance with transport tempo. The
snap-sequencer has many editing modes that allow for modifying the snapshots content (snap, amp, etc..).
• output
The snap-sequencer outputs a list of data each time a step is triggered. Each data type (snap, amp, amp
_rand...) is interpreted in order to recall a snapshot, with additional possibilities such as: random amplitude of
envelope and snap recall probability.
46
Up to eight time intervals can be selected. The choice is performed with a random
function, according to a weight probability for each possible interval - an interval with a
weight of 2 will have twice as many chances to be selected by the random procedure that
an interval that has a weight of 1.
You can set "general changes probability" (75 % in the example) , 0 % no interval
change occur, as all intervals probability weight to 0. Change and no change led (yes/
no), shows if the random parts it’s to change or not. Random number generator fall every
“change interval” time (2 in the example), beat sequence is displayed on the right.
The most left reset button, resets
sequencer to first step, in
permutation mode reinit
sequence, no action in random mode.
Edit mode selection is done using this set of tabs, view all displays all of the available data (and allows you
to edit the snapshots). Snap mode displays and allows editing of the snap only. Amp mode allows you to
edit the volume of snapshot envelope. Duration mode is used to change the step size of each value, while
Probability mode allows you to manipulate the likelihood of a snapshot.
Sequencer play mode menù read steps in the explained way. You can chose
the number of the step (columns) and snapshots range (row), then fold/
unfold lets you chose to display all possible snapshots, or only a specific set
of snapshots.
Some kinds of transformations are available their names are auto-explicative.
Through the four arrow keys at the bottom, up and down (snap number) or left and right (in time) you can
shift the entire sequence.
Snapshots Sequencer is an ADSR (Attack Decay Sustain Release) every step envelops the streams (or
main) amplitude audio signal.You can define a Attack Decay time and Sustain percentage, at last Release
time. When Sequencer is turned off, audio signals (streams or main) pass through normally.
• global transport
47
The global transport it’s designed to access quickly all what is needed to control the clock of timing Pulsaret
objects.
Here is a listing of the note and tick values associated with common note durations. Note value abbreviations
that can be used in Pulsaret to specify time are in bold.
128n - One-hundred-twenty-eighth note - 15 ticks
64n - Sixty-fourth note - 30 ticks
64nd - Dotted sixty-fourth note - 45 ticks
32nt - thirty-second-note triplet - 40 ticks
32n - thirty-second note - 60 ticks
32nd - Dotted thirty-second note - 90 ticks
16nt - Sixteenth note triplet - 80 ticks
16n - Sixteenth note - 120 ticks
16nd - Dotted sixteenth note - 180 ticks
8nt - Eighth note triplet - 160 ticks
8n - Eighth note - 240 ticks
8nd - Dotted eighth note - 360 ticks
4nt - Quarter note triplet - 320 ticks
4n - Quarter note - 480 ticks
4nd - Dotted quarter note - 720 ticks
2nt - Half note triplet - 640 ticks
2n - Half note - 960 ticks
2nd - Dotted half note - 1440 ticks
1nt - Whole note triplet - 1280 ticks
1n - Whole note - 1920 ticks
In Pulsaret you can find also this formula of value.
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Hyper Vectorial Pad1
• general explanation
HV_pad (i.e. Hyper Vector Pad) is 2D parameter interpolation user interface. Displays nodes in a 2dimensional space, and calculates the distance from a pointThe distance factor determine the weight
between snapshots interpolation. Like for snapshots-module, we have ten HV_Pad grouped in a unique
window.
• main;
• streams 1÷8;
• gsnap;
1
See VMCI Virtual Midi Control Interface (http://www.csounds.com/maldonado/), thanks Gabriel Maldonado.
49
You can open HV_Pad window from application menù bar (cmd + P on Mac or ctrl + P on Windows) or
through the tool bar button.
The vectorial pads (10) are contained inside HV_Pad: main, streams (8) and global snapshots, clicking on
tab you change among them.
HV_Pad extends the functions of micro-pad, interpolating until 24 snapshots together, thus you can modify
hundreds of parameters with a single mouse motion, according to a structure configured by the user. This
control method is called Hyper Vectorial Synthesis Control. Normally the user interacts with these areas with
the mouse or through self-scanning exploration, however a remote MIDI control is possible, by assigning the
X Y pointers (see MIDI/OSC I/0).
In the image below, from number box (red circle) you can sets
the number of nodes, you can set a specific configuration
(system nodes) by clicking and dragging center of node to
move it or Option Key (alt) to change node size.
N.B. The nodes are numbered progressively,
therefore each node (on the pad) must be
corresponds by a stored preset snapshot. If you
50
employ an empty snapshot, the interpolation between them will not work properly.
Also you cannot disabling completely the nodes (node area = 0), to avoid fall in the
same problem.
• pad mouse behavior
The pop-up menù allow you to chose mouse behavior:
•Automatic: allows the mouse to change the nodes position,
size and slider (pad scrub);
•Nodes: the mouse can only edit the nodes position and size,
alt + mouse scroll to change node size; click and drags on
the node number, to move it;
•Slider; the mouse can only changes the slider location (pad
scrub).
You can automatize HV_pads scrub (slider) choosing a scanning mode:
Scanning method:
reflects causes a reflection when the cursor reaches the edge;
jitter outputting random numbers within a moving specified range around current scrub mouse position;
drunk will perform a "drunk" walk, creating unpredictable paths;
spiral centripetal, centrifugal or palindrome, with phase deformation parameter;
The jitter step size (number box 8) it will be showed only
when jitter is selected. This parameter sets the random
amount of jittering around of the mouse position during
scrub pad.
Automatic exploration in reflects, jitter and drunk, you can interact with the pad. The values of mouse and
direction, will be the new coordinates of scanning. Instead in spiral you cannot interact with pad
• X/Y time
The X Y indexes scanning ratio, depends by the "frequency" X/Y axes. These are controlled by the two
sliders showing value in seconds (i.e. time = 1/frequency).
51
You can set scanning time for the two axis (2D XY). For example 3.50 sec. for the X axe it means that
horizontal slider scrub it take 3.50 sec to pass from left edge to right edge of the pad, Y = 5.00 sec.
X Y parameters, they assumes different means according to selected scanning mode:
in reflects is expressed in time, is the time it takes to move from X = left/right and Y = up/down;
in drunk is expressed in time, is the deviation amount rnd(X) and rnd(Y);
in spiral: X is the time for each sub-spiral, Y is the total duration until reach all pad area;
The spiral mode, will enable two related parameters:
centripetal moves from outside to inside, when it reaches the limit again, repeat;
centrifugal does the opposite;
palindrome is a combination of both;
The spiral phase deformation (number box 0.25) 0: no spiral; 0.25 circular; > 0.25 crushed; > 0.5 clockwise
rotation
• miscellaneous
Auto-exploration is possible only when the DSP is on (see DSP_settings),
on/off toggle auto-exploration according to selected mode.
The red circled number box, sets the interpolation update time in milliseconds, this is a speed limit to avoid
an overhead of cpu with complex interpolations, be careful with short times!
A little status bar at the bottom of HV_Pad, show the currents pads enabled for auto
scanning. In fact passing among ten available pads, we need a monitor remembering us
pads at works.
52
Matrix
• general explanation
Pulsare.m4l matrix linkage is employed to create a dependency between parameters. Moreover you can
rescale, reverse controlled parameter range and/or move it by LFO (Low Frequency Oscillator). This
technique is named: Grainlet Synthesis.
• parameter linkage
On the top of the Matrix (horizontal) we have sending parameters, They will be linked with right ones
(vertical). moving a send parameter, you will move controlled (linked) parameter.
Every controlled parameter can be mapped on a own range and/or reversed, it means it will be moved on the
contrary regarding the motion of the parameter that controls it.
53
• grainlet/pulsaret mode
In grainlet mode you you can interconnects all granular parameters freely, you can also toggle in Pulsaret
mode.
A Pulsaret consists in a brief burst of energy. The grain duration contains only one cycle, varying the grano
length you change “duty cycle” of pulsaret, while density remain fix and indipendent. You can link the two
parameters in two ways: length -> cps or cps -> length.
cps = (1/length) * cycleN
length = (1/cps) * cycleN
e.g. 100 Hz = 1/100 = 0.01 ossia 10 ms.
Optionally you can specify the number (numberbox) of cycles for the pulsaret.
e.g.
cycleN = 2
cps = 100 Hz
length = 1/100 = 0.01 * 2 = 0.02 Sec (ossia 20 Ms)
for the cps/length linkage:
cycleN = 3
length = 45 Ms = 0.045 sec
cps = 1/0.045 * 3 = 66.66 Hz
• parameters rescale
54
In the below example, length and cps are linked:
all length range is employ to move cps parameter.
If can maps entire length range on a little part of cps parameters:
You can also enter your values on the numbox (27.5 - 16.000), for changing global range and change
exponential curve mapping (numbox):
- min value for the selection (click and digit a new value, then press enter)
- max value for the selection (click and digit a new value, then press enter)
- exponential base value (default 1. = linear)
If pickup is enable, when controlled parameter is out of range, it will no moved until it will be bring inside of
the range. it wock in analogous way to MIDI/OSC pickup, see MIDI/OSC for more details.
same dir/inverse, make an inverse maps of the controlled parameter.
the last parameter, controlling general rescale output, 1. it means no rescale.
• LFO modulation
Low Frequency Oscillator (LFO) is usefull to automate one or more parameters, they will be constantly
moved around one middle value depending by shape, frequency and deep of modulation of LFO.
55
in the above example, LFO1 (of the 5 availables) modulate volume parameter, LFO2 panning etc.... After
you have assign matrix linkage, you need enable LFO clicking on the on button.
The LFO params are (from top to bottom):
-on/off LFO (Low Frequency Modulation)
-toggle between beat synced and hz; in sync mode, the beat division switches specify the
delay time in 16th notes; in time mode, the pan rotation is expressed in hz.
-panning rotate frequency in Hz
-16 note: each numbered switch represents the time in 16th notes, note that these switches
are only active if the time is set to sync; you can change global time by the transport unit.
-lfo shape, you can select one of the windowing shapes availables
-modulation deep in %
-uni/bipolar range, unipolar = 0 to 1 modulation range (with 100% deep); bipolar= -1 +1
-change the speedlimiting time parameter update.
• status bar
56
In the Pulsaret.m4l status bar, report current matrix
modulation state.
The receiving parameter wait until sending parameter
reach the same value i.e. pickup feature, (Ableton 1 Live
inspired).
Pickup system wait until sending parameter reaches
receive widget position. On the status bar you will see a
message shown the gap between send/receive
parameters.
On the most left status bar we have grain overlaps
factor, you need enable tiny orange button.
1
Ableton Live is a loop-based software music sequencer and DAW for Mac OS and Windows by Ableton. The latest major release of
Live, Version 8, was released in April 2009. In contrast to many other software sequencers, Live is designed to be an instrument for live
performances as well as a tool for composing and arranging. It is also used for mixing of tracks by DJs, as it offers a suite of controls for
beatmatching, crossfading, and other effects used by turntablists, and was one of the first music applications to automatically beat
match songs. It does not support traditional musical notation. (Wikipedia)
57
MIDI/OSC I/O
• header
Open MIDI/OSC I/O from application menù bar CMD + M (Mac) or CTRL + M (Win) or from tool bar, the
header is divided in two parts: MIDI (Music Instrument Digital Interface) and OSC (Open Sound Control). You
can also browse quickly the parameters-families through the below menù:
• gain~: mixer granular streams faders;
• play/stop: mixer granular streams triggers;
• solo: mixer solo toggles;
• mute: mixer mute toggles;
• parameters: granular streams parameters;
• shared: common controllers;
MIDI:
Pulsaret routing only Cc (Control Change) MIDI messages, no Pc (Program Change), no Note On/Off, no
Poly Pressure, no After Touch and other MIDI messages are supported. The Control Change messages,
generally send continuous data between 0 ÷ 127 (i.e. 7 bit) steps resolution. You can enable 14 bit supports
to improve resolution (see below).
For motorized devices, Pulsaret can synchronize software parameters with the hardware controllers. Thus
when you change, for example, number of stream from GUI, Pulsaret send value through MIDI and OSC
output ports. You will have software and hardware parameters synchronized.
I/O port selecting:
The two pop-up menùs shows a list of available ports according to hardware device detected, click refresh
to update list, useful when hot connecting a device(s).
Below the MIDI I/o ports, we have “MIDI monitor” where you can see raw coming MIDI data. Midi monitor
interpret raw MIDI data parse, providing a short description about generic messages kind and MIDI monitor
is able to understand if incoming message are 7 or 14 bit resolution.
OSC:1
Dispatch messages through an OpenSound Control address hierarchy with pattern matching. Values are
serialized and sent over the network as OSC compatible UDP;
Input port (3009) set input port to send messages to at host;
Output port (1974) set output port to send messages;
1
OSC is a CNMAT Max objects, can be found at: http://www.cnmat.berkeley.edu/MAX
58
Tiny oranges buttons enable/disable respective OSC I/O.
Very important!: default values for I/O ports: 30-09-1974 is the author birthday date.
We need also configure host (127.0.0.1) output , entering an IP address or hostname, clicking on localhost
to assign default ip value (i.e. local host 127.0.0.1).
Like MIDI monitor, OSC monitor show OSC I/O data, you can select with mouse the text string address
(OSC path) and paste it on widget controller (see below).
You can enter a prefix ID (can be numbers and/or text) for the the OSC output address. When an incoming
OSC message is synced on a widget parameter, this read and write values employing the same path
address. Some time is useful append an id prefix to differentiate path in order (for instance) to send message
to another device .
N.B. OSCulator 1 can be employed like MIDI/OSC monitor beyond supplying advanced features.
N.B. All settings will be saved on the project (see overview for more explanations)
• sync
Sync system allow you synchronize software parameters with hardware devices. Thus Pulsaret send out
through MIDI and OSC ports parameters data. Default sync is disabled for all widgets, then no MIDI/OSC
data are sent. You can select enable sync out for each widget controller or superimpose a common feature
at all widgets (see below).
For example, you can enable sync for all widgets controllers, by selecting the item from header menù.
Sync have five outputs mode:
MIDI/OSC out mode for all widgets:
•off no MIDI/OSC outs data;
•down send MIDI/OSC data when click-down left mouse button;
•up send MIDI/OSC data when click-up left mouse button;
•downup send MIDI/OSC data when click-down and click-up;
continuous
send MIDI/OSC data continuously. N.B. this overwrite all settings of the
•
widgets.
Every time you move a Pulsaret parameter on the GUI or when a transition/sequencer/HV_pad is at work,
sync values are sent.
To avoid an overhead of outputs data, through the number box (150 ms) you can change MIDI/OSC output
update rate in milliseconds. For instance during the snapshots transitions, you should limiting sync data in
order to preserve a normal cpu overhead: MIDI out update rate, expressed in milliseconds set the speed
limit MIDI output data.
• widget mapping
1
OSCulator is a software that can be used with many different hardware devices and software. For example, with OSCulator, your
Nintendo Wiimote can talk to major MIDI sequencers or your favorite console emulator or even the Kyma sound design workstation.
And your iPad, iPhone, iPod Touch or Lemur can do just the same… with great ease of use. http://www.osculator.net
59
We have two widgets mapping kinds:
1. Cc_raw
2. Cc_scale
Cc_raw is addressed to mapping Pulsaret buttons/toggle parameters, Cc_raw read/write MIDI/OSC
returning integer values (between 0 and 127 for MIDI), Learn function waits to feel a MIDI CC or OSC
address, when it happens the the control is set up on the incoming CC number and MIDI channel or OSC
path address.
Now you can controlling Pulsaret parameter via MIDI and/or OSC.
Current MIDI Cc, Channel number and raw value are shown. Of course you can manually change MIDI Cc or
channel. N.B. sets Cc to -1 for disabling controller.
Both MIDI and OSC have learn feature. Toggle learn and send a MIDI or OSC data in order to assign
incoming Cc or OSC address. In the example, Cc 23 on first MIDI channel and OSC /play/stop[1] address
are assigned to play/stop[1] Pulsaret parameter, if sync is enabled Pulsaret also is able to send MIDI/OSC
data. It is possible change OSC path value entering a new one.
An OSC message can be a list, if you want read an element of the list, you can select an OSC input item
from the list, (default first element). In the example, Nintendo WII device send the accelerometer values like
a three list items, selecting 3 to read only third incoming.
I/O sync pop-up menù, set output mode for current widget, see sync for better understand.
The most right menù (toggle), set the controller kind:
• toggle switch from 1 and 0 every time MIDI/OSC I/O == 0;
• button send 1 every time MIDI/OSC input != 0;
• incr and decr increment or decrement (regardless of direction) by one step, every time MIDI/OSC input ==
0. when values 127, or 0 are reached, the next value is wrapped;
• incr/decr it counts upward until it reaches 127, then counts down until it reaches 0, then up, then down,
and so on;
• controller mode read raw data from MIDI/OSC input, raw data MIDI between 0 and 127.
Cc_scale extends Cc_raw functions, you can choose a certain range on which rescale the parameter, for
example control gain1, which is associated, covers a range -70 ÷ +6 (76 units). You know that MIDI has 7 bit
resolution covering 128 steps range units (0 ÷127).
In the example, the Cc 2 is scaled on a part of the entire parameter range (11.36) by the orange horizontal
selection. You can also change min/max values for the parameter (box numbers -70. and 6.), current range
will be reported.
60
OSC assumes default values between 0 and 1. You can also change OSC input range. For example we
control gain1 through /wii/1/accel/roll OSC data incoming from OSCulator and Nintendo WII, we rescale the
range for controlling only a middle part of parameter.
Pulsaret allow 14 bit precise MIDI input controller. This feature is
achieved by union of two controls change on the gap by 32. For instance
if midi learn finds Cc1: 7, the second Cc2 it will be 39 (i.e. 7 + 32 = 39).
In other words, Cc1 is the most-significant byte controller number (1-127), while Cc2 is the least-significant
byte controller number (1-127). Activating 14 bit, the Cc2 number it will be shown.
N.B. when you send 14 bit precise controller, Cc1 it should not be greater than to 95, this is because the Cc2
it should be not greater than to 127, i.e. 95 + 32 = 127.
Inv, invert the I/O range for the widget.
Most right menù, I/O sync (off) set the output mode for current widget, see sync for better understand and
numbox (1.00) on top right set the exponential base value (default 1. = linear). The number is converted
according to the following expression: y = b e-a log c ex log c.
N.B. sets Cc to -1 for disabling controller.
• stream select
Having Pulsaret eight granular streams, we have an huge number of parameters to be controlled. Therefore
it would be difficult to manage all parameters independently. An simple idea could be to assign each stream
on own MIDI channel, for example stream1 -> MIDI channel 1, stream 2 -> MIDI channel 2 and so on... but
each parameters of the eighth streams should have the own MIDI/OSC widget. To avoid this, Pulsaret can
61
be controlled following a layout strategy. MIDI Cc or OSC messages controlling only selected granular
stream
(i.e. stream GUI in foreground).
The streamsel widget (first on the top), is crucial because it allows to bring the stream in foreground to be
controlled via MIDI/OSC.
MIDI/OSC I/O works only on current selected granular stream (foreground). You need bring in foreground
a granular streams, in order to controlling via MIDI/OSC their parameters.
Disabling auto mode (manual top left toggle), a menù tab will be shown allow you to chose a granular
stream. Thus when you change granular stream (from main GUI), MIDI/OSC controls always the same
stream. When auto is enabled, the granular streams in foreground(selected from GUI tab), will be controlled
via MIDI/OSC.
• shared
Through this section you can controlling some shared parameters like: windowing gauss/curve
deformation, snapshots recall, snapshots time transition and HV_pad X/Y.
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Like for the parameters section we need selects a granular stream where send MIDI/OSC data, also this
Pulsaret parameters follows same modus operandi. For example every snapshots module: main, streams
(8) and gsnaps, (total 10) can be controlled by main-stream-gsnap-switch widget.
Idem for HV_ Pads.
The first three widgets are assigned to Windowing shapes: gauss and curve, see Windowing module for
more details, they refers to gauss and curve shapes deformations.
The central switch tab or main-stream-gsnap-switch widget, pass the control to relative section.
When stream is selected, you will see a number according to selected stream.
Like streamsel, you can controlling main-stream-gsnap-switch via MIDI/OSC.
• pick-up
When sync is disabled, the pickup feature, (Ableton1 Live inspired) is automatically enabled.
If the incoming MIDI/OSC value is different from Pulsaret GUI parameter value, pickup system wait that
MIDI/OSC value reaches the widget. On the status bar you will see widget and incoming MIDI/OSC values
until one of them does not match.
Enabling sync mode, will disable pickup system.
N.B.
In order does not disable pickup system, you should disable sync out when you have not a motorized MIDI
controller, (like Behringer BCF2000).
1
Ableton Live is a loop-based software music sequencer and DAW for Mac OS and Windows by Ableton. The latest major release of
Live, Version 8, was released in April 2009. In contrast to many other software sequencers, Live is designed to be an instrument for live
performances as well as a tool for composing and arranging. It is also used for mixing of tracks by DJs, as it offers a suite of controls for
beatmatching, crossfading, and other effects used by turntablists, and was one of the first music applications to automatically beat
match songs. It does not support traditional musical notation. (Wikipedia)
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Overview
• save/load project
Project management is maybe, the most important Pulsaret feature. The idea is to capture and save all
Pulsaret settings like: snapshots presets, MIDI/OSC mapping, setup configurations, sound files and sound
files informations etc... in a single project i.e. a folder containing everything.
Open... allow you to browse filesystem to chose the project folder, N.B. you need select entire folder. You
cannot select subfolders or any files contained in the project folder but from filesystem browser you need
simply select the project folder.
N.B. Alternatively you can drag and drop project folder on the main mixer.
Save/Save As... create new project if not saved yet or save current project with name.
When you save a project without consolidate audio files, you will not able to open project correctly on the
others computers. This because stored sampled files paths, refers to local path. Thus also if you change
sampled files path on your machine, you will not able read correctly project. Alternatively consolidate
feature, follow explained, solve the problem.
At each files exported is attached a prefix ID, the ID is shape for Windowing module and stramN for
granular streams, where N is the slot number (12). All exported sampled files are in AIFF format, so .aiff
extension will be attached eventually.
e.g
stream3snd2filname.mp3.aiff
for streams or
shapesnd12filname.wav.aiff
for windowing module
Consolidate, if enabled when you Save or Save AS... project, all sampled sound files loaded on the
streams slots (12 available), LIVE buffer contents
and/or Windowing sampled shape slots (12
available), they will be exported inside project folder. e.g. if enabled the files will be exported on the
consolidate folder inside project folder, if disabled the files will be loaded from filesystem path.
Useful If you intend open projects everywhere.
N.B.
• You cannot use \ or / symbol in the project name folder;
• When you Save newly the project with consolidate flag off, the originals sampled files
path will be restored but files stored in consolidate folder they will not deleted.
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• project tree
In main.xml are stored all general configuration and the snapshots presets for main section;
In globalsnaps.xml are stored the gsnap global presets;
streamN.xml contains the streams snapshots presets, where N is the granular stream number;
streamN.txt contains granular streams data collector like path of sound files loaded, general information
such: zoom, normalize, length selection, file sample length, sampling rate, time duration etc...
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windowing.txt is collector file for the Windowing module: path of sound files loaded, general information
such: zoom, normalize, length selection, file sample length, sampling rate, time duration etc...
TextNotes.txt is where Pulsaret save your note.In fact Pulsaret provide a text editor where, for instance you
can write “the live set performance score” or what you want. When you save project also notes are included;
vst folder contains .fxp bank files for each allocated vst slot. In the example only the ninth (MASTER) slot,
hosts a VST plugin. Instead vst.txt contains general information about loaded plugins: name and path;
consolidate folder contains all consolidated audio files, exported when you save (if consolidate is enabled)
your project. See above for more details;
• status bar
Pulsaret status bar is subdivided in three parts, from left (frame blu) we have main status bar where are
displayed general DSP infos like sampling rate, I/O vector size, signal vector size, cpu utilization.
In order to show cpu utilization value you need toggle tiny orange button.
N.B. You can change from status bar only Sampling Rate, see DSP Settings for explanations.
The current sampling rate will be saved on the Pulsaret project, this because if you have consolidated sound
files in the project, it’s important load the project with own SR. i.e. The project Sampling Rate and exported
consolidated files must have a coherent sampling rate.
On the right (purple frame) we have the pick-up display. See pick-up for details
Two led green circled, displays the global transport beat and MIDI/OSC activity. You can also open their
window by clicking.
On the right part of above image, yellow framed, we have granular streams information, these values refer to
current selected granular stream.
Mainly, Granular stream status bar, grain overlaps factor (on the left) and WavePad sampled file informations
like: current file name, current file length in sample, current file sampling rate, current file length in
milliseconds. In order to enable grains overlap report, you need enable tiny orange button.
• clue window
To avoid annoying hint messages when you are working on the GUI, Pulsaret sends all hints in the clue
window. In the clue window you can read some informations about Pulsaret GUI widget, when the mouse is
over it.
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The clue window CMD + I (Mac) CTRL + I (Win) is a floating window that displays documentation when you
bring mouse pointer on a widget. You can use it to obtain reference information for objects then close
window during performance.
• float window
Have you noticed almost all Pulsaret windows contains in a corner (generally top right) a tiny
orange button? toggle it to set float/no float window. When toggle float, current window will be
always in front. This feature is saved on project like general setup.
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• history
ver 1.0 (november 2010) official release built with Max/MSP version 5.1.5
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eight granular streams simultaneously
envelope/windowing menage up to 16 pre-generated shape (prototypes)
envelope/windowing loading and menage up to 12 sound-files (aiff,wav,mp3)
dynamic envelope buffer load/save, normalize, trim, resize length etc...
main mixer 8 channels + 1 master, solo/mute and VST slots, Master cascade~ filter
multichannel I/O mapping
quick-record export master channel and/or multichannel file streams, progressive file autoname, select
directory and re-sampling/quantize out file
snapshots (presets) memory: up to 100 for each stream and 10 for the main mixer and global snapshots
simultaneously (streams and main) transition (interpolation) between snapshots (in a given time)
clients menage: include/exclude widgets from transitions
four draw table for transitions curves
micro-pad interpolating between four snapshots
HV_pad (i.e. hyper vectorial pad), 9 snapshots pad (4 pad near), and auto-explorer (spiral, dunk, reflects)
engine
MIDI/OSC input mapping: learn/manual, rescale range and exponential curve
MIDI/OSC output sync: enable/disable, continuously or mouse up send
fully managing the project (as a folder), save/save as and load
drag and drop on windowing wave-pads (audio files or folder) to fill menù soundfiles or project folder in
main windows to load project
stereo panning rotation: pan, jitter, lfo (shape/hz)
panning time, note based sync.keyboard frequency select (streams)
windowing sound files consolidate (copy buffers in the prj folder)
every stream can deforming own windowing envelope (attack/decay)
global presets, menage all Pulsaret widgets together
snapshots list, consecutive rename
global transport time
snapshots sequencer rhythms improviser unit
windowing deformation shape MIDI/OSC
clue windows, report widgets information under the mouse
matrix parameter linkage, rescale and LFO modulation range in Grainlet Synthesis
Pulsaret length/cps and viceversa dependency
cascaded series of biquad filters for each stream
simply signal-oscilloscope, that allows you to monitor the visual progression of master waveform
ver 2.0.0 (april 2011)
new GUI lock
new message box does not stop audio
TOOLS BAR quick functions access
HV_pads 2D nodes parameters interpolation
HV_pads add jitter exploration mode
HV_pads "reflects", "jitter" and "drunk", now you can interact with the pad. The values of mouse and
direction, will be the new coordinates of scanning
• HV_pads on/off monitor in main window
• MIDI/OSC 14 bit precise Control Change Midi I/O
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MIDI removed pich bend in favour of 14 bit MIDI I/O
MIDI/OSC new midi monitor, parse 7 and 14 bit control change and show detailed messages
MIDI/OSC OSC learn function
MIDI/OSC you can append a prefix ID on the OSC Output adress
MIDI/OSC OSC I/O send and receive automatically in the same adress
MIDI/OSC improved rescale range (old project are't compatible, sorry)
MIDI/OSC MidiRaw send Midi/Osc data out for sync
MIDI/OSC MidiToggle integrated in MidiRaw OK
MIDI/OSC new MIDI/OSC sync mode: mouse down, up, downup, continuous
MIDI/OSC add MidiRaw input data mapping incr, decr, incr/decr modes
MIDI/OSC MIDI CC = -1 (default) disable all MIDI I/O widgets
MIDI/OSC toggle switch now show correctly own state
MIDI/OSC you can chose an item by a list of four elements like OSC input
MIDI/OSC inverse mapping for MIDI/OSC I/O
MIDI/OSC OSC void string bug fixed
GSNAPS defaults all turnoff DSP
NOTES add you can save your notes in the project
STREAMS PARAMETERS moltiply/divide (/4 /2 x2 x3 x4) feature
GUI faders relative/absolute mousing mode switch
panning now update correctly value when you select manual
WAVEPAD shortcuts (F1, F2, F3, F4) enabled
LIVE/Sample button switch
snap-seq add global transport trigger link enable/disable
MIXER native FILTER 24 band equalizer channels and Master
MIXER native reverb channels and Master
MIXER native compressor fchannels and Master
MIXER rebuilded solo/mute
MIXER add solo/mute fade in/out
FILTER add filters types select
FILTER add cutoff, gain and Q/S fine controller
FILTER add dry/wet
FILTER dynamic filter allocation, until 24
FILTER dry/wet controller
FILTER auto generate filter bands: Harmonic, Geometric, Scalar and Fibonacci
FILTER frequency zoom in/out
WINDOWING/WAVEPAD unsupported files does't overwrite path bug fixed
Record Master/Multichannels save file/folder patch in project
Record Master/Multichannels new GUI lock
New detailed user manual online
SNAP-SEQ step jump bug fixed
SNAP-SEQ add portamento transition
SNAP-SEQ some gui adjustments
MATRIX improved rescale range (old project are't compatible, sorry)
OSCILLOSCOPE stereo
WINDOWING additive jump hrm amp select, fixed
HV_Pad optimized auto scanning (drunk)
HV_Pad save on the project, the last pointer position
clue windows show now the widgets name correctly
HV_Pad drunk/spiral/reflect/jitter cpu improved performance
MIDI/OSC correct pan range rescale
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FILTER improved and refined
FILTER enabled for transitions
GSNAPS micropad became 2D nodes
GSNAPS speedlimit transition update
new micropad 2D-nodes
• acknowledgments
I would like to thanks Eugenio Giordani and Nicola Casetta, a special thanks to: Renato Alberti, Felix
Petrescu and Pasquale Ascione.
Pulsaret.m4l Max externals and abstractions:
© Eric Lyon el.Granule~ modified by ape 2011 in ape.Granule~
http://www.gmem.org/
All rights reserved
CNMAT ©
OpenSoundControl (OSC-route)
http://www.cnmat.berkeley.edu/MAX
All rights reserved
yafr2, plate reverb, in the style of Griesinger. Randy Jones [email protected]
Pulsaret was created with Max5
http://www.cycling74.com
The Author:
Alessandro Petrolati
apeSoft
All rights reserved © 2010-2011
[email protected]
www.densitygs.com
www.alessandro-petrolati.it